3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
31 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
68 #define RTSP_MEDIATYPE_OPTS(name, longname) \
69 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
70 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 const AVOption ff_rtsp_options[] = {
75 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
76 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
77 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
78 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
79 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
81 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
82 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
83 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 static const AVOption sdp_options[] = {
88 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
93 static const AVOption rtp_options[] = {
94 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
98 static void get_word_until_chars(char *buf, int buf_size,
99 const char *sep, const char **pp)
105 p += strspn(p, SPACE_CHARS);
107 while (!strchr(sep, *p) && *p != '\0') {
108 if ((q - buf) < buf_size - 1)
117 static void get_word_sep(char *buf, int buf_size, const char *sep,
120 if (**pp == '/') (*pp)++;
121 get_word_until_chars(buf, buf_size, sep, pp);
124 static void get_word(char *buf, int buf_size, const char **pp)
126 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
129 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
131 * Used for seeking in the rtp stream.
133 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
137 p += strspn(p, SPACE_CHARS);
138 if (!av_stristart(p, "npt=", &p))
141 *start = AV_NOPTS_VALUE;
142 *end = AV_NOPTS_VALUE;
144 get_word_sep(buf, sizeof(buf), "-", &p);
145 av_parse_time(start, buf, 1);
148 get_word_sep(buf, sizeof(buf), "-", &p);
149 av_parse_time(end, buf, 1);
151 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
152 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
155 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
157 struct addrinfo hints, *ai = NULL;
158 memset(&hints, 0, sizeof(hints));
159 hints.ai_flags = AI_NUMERICHOST;
160 if (getaddrinfo(buf, NULL, &hints, &ai))
162 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
168 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
169 RTSPStream *rtsp_st, AVCodecContext *codec)
173 codec->codec_id = handler->codec_id;
174 rtsp_st->dynamic_handler = handler;
175 if (handler->alloc) {
176 rtsp_st->dynamic_protocol_context = handler->alloc();
177 if (!rtsp_st->dynamic_protocol_context)
178 rtsp_st->dynamic_handler = NULL;
182 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
183 static int sdp_parse_rtpmap(AVFormatContext *s,
184 AVStream *st, RTSPStream *rtsp_st,
185 int payload_type, const char *p)
187 AVCodecContext *codec = st->codec;
193 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
194 * see if we can handle this kind of payload.
195 * The space should normally not be there but some Real streams or
196 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
197 * have a trailing space. */
198 get_word_sep(buf, sizeof(buf), "/ ", &p);
199 if (payload_type >= RTP_PT_PRIVATE) {
200 RTPDynamicProtocolHandler *handler =
201 ff_rtp_handler_find_by_name(buf, codec->codec_type);
202 init_rtp_handler(handler, rtsp_st, codec);
203 /* If no dynamic handler was found, check with the list of standard
204 * allocated types, if such a stream for some reason happens to
205 * use a private payload type. This isn't handled in rtpdec.c, since
206 * the format name from the rtpmap line never is passed into rtpdec. */
207 if (!rtsp_st->dynamic_handler)
208 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 /* search into AVRtpPayloadTypes[] */
213 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
216 c = avcodec_find_decoder(codec->codec_id);
222 get_word_sep(buf, sizeof(buf), "/", &p);
224 switch (codec->codec_type) {
225 case AVMEDIA_TYPE_AUDIO:
226 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
227 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
228 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
230 codec->sample_rate = i;
231 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
232 get_word_sep(buf, sizeof(buf), "/", &p);
236 // TODO: there is a bug here; if it is a mono stream, and
237 // less than 22000Hz, faad upconverts to stereo and twice
238 // the frequency. No problem, but the sample rate is being
239 // set here by the sdp line. Patch on its way. (rdm)
241 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
243 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
246 case AVMEDIA_TYPE_VIDEO:
247 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
249 avpriv_set_pts_info(st, 32, 1, i);
254 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
255 rtsp_st->dynamic_handler->init(s, st->index,
256 rtsp_st->dynamic_protocol_context);
260 /* parse the attribute line from the fmtp a line of an sdp response. This
261 * is broken out as a function because it is used in rtp_h264.c, which is
263 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
264 char *value, int value_size)
266 *p += strspn(*p, SPACE_CHARS);
268 get_word_sep(attr, attr_size, "=", p);
271 get_word_sep(value, value_size, ";", p);
279 typedef struct SDPParseState {
281 struct sockaddr_storage default_ip;
283 int skip_media; ///< set if an unknown m= line occurs
286 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
287 int letter, const char *buf)
289 RTSPState *rt = s->priv_data;
290 char buf1[64], st_type[64];
292 enum AVMediaType codec_type;
296 struct sockaddr_storage sdp_ip;
299 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
302 if (s1->skip_media && letter != 'm')
306 get_word(buf1, sizeof(buf1), &p);
307 if (strcmp(buf1, "IN") != 0)
309 get_word(buf1, sizeof(buf1), &p);
310 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
312 get_word_sep(buf1, sizeof(buf1), "/", &p);
313 if (get_sockaddr(buf1, &sdp_ip))
318 get_word_sep(buf1, sizeof(buf1), "/", &p);
321 if (s->nb_streams == 0) {
322 s1->default_ip = sdp_ip;
323 s1->default_ttl = ttl;
325 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
326 rtsp_st->sdp_ip = sdp_ip;
327 rtsp_st->sdp_ttl = ttl;
331 av_dict_set(&s->metadata, "title", p, 0);
334 if (s->nb_streams == 0) {
335 av_dict_set(&s->metadata, "comment", p, 0);
342 codec_type = AVMEDIA_TYPE_UNKNOWN;
343 get_word(st_type, sizeof(st_type), &p);
344 if (!strcmp(st_type, "audio")) {
345 codec_type = AVMEDIA_TYPE_AUDIO;
346 } else if (!strcmp(st_type, "video")) {
347 codec_type = AVMEDIA_TYPE_VIDEO;
348 } else if (!strcmp(st_type, "application")) {
349 codec_type = AVMEDIA_TYPE_DATA;
351 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
355 rtsp_st = av_mallocz(sizeof(RTSPStream));
358 rtsp_st->stream_index = -1;
359 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
361 rtsp_st->sdp_ip = s1->default_ip;
362 rtsp_st->sdp_ttl = s1->default_ttl;
364 get_word(buf1, sizeof(buf1), &p); /* port */
365 rtsp_st->sdp_port = atoi(buf1);
367 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
369 /* XXX: handle list of formats */
370 get_word(buf1, sizeof(buf1), &p); /* format list */
371 rtsp_st->sdp_payload_type = atoi(buf1);
373 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
374 /* no corresponding stream */
376 st = avformat_new_stream(s, NULL);
379 st->id = rt->nb_rtsp_streams - 1;
380 rtsp_st->stream_index = st->index;
381 st->codec->codec_type = codec_type;
382 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
383 RTPDynamicProtocolHandler *handler;
384 /* if standard payload type, we can find the codec right now */
385 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
386 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
387 st->codec->sample_rate > 0)
388 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
389 /* Even static payload types may need a custom depacketizer */
390 handler = ff_rtp_handler_find_by_id(
391 rtsp_st->sdp_payload_type, st->codec->codec_type);
392 init_rtp_handler(handler, rtsp_st, st->codec);
393 if (handler && handler->init)
394 handler->init(s, st->index,
395 rtsp_st->dynamic_protocol_context);
398 /* put a default control url */
399 av_strlcpy(rtsp_st->control_url, rt->control_uri,
400 sizeof(rtsp_st->control_url));
403 if (av_strstart(p, "control:", &p)) {
404 if (s->nb_streams == 0) {
405 if (!strncmp(p, "rtsp://", 7))
406 av_strlcpy(rt->control_uri, p,
407 sizeof(rt->control_uri));
410 /* get the control url */
411 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
413 /* XXX: may need to add full url resolution */
414 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
416 if (proto[0] == '\0') {
417 /* relative control URL */
418 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
419 av_strlcat(rtsp_st->control_url, "/",
420 sizeof(rtsp_st->control_url));
421 av_strlcat(rtsp_st->control_url, p,
422 sizeof(rtsp_st->control_url));
424 av_strlcpy(rtsp_st->control_url, p,
425 sizeof(rtsp_st->control_url));
427 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
428 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
429 get_word(buf1, sizeof(buf1), &p);
430 payload_type = atoi(buf1);
431 st = s->streams[s->nb_streams - 1];
432 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
434 } else if (av_strstart(p, "fmtp:", &p) ||
435 av_strstart(p, "framesize:", &p)) {
436 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
437 // let dynamic protocol handlers have a stab at the line.
438 get_word(buf1, sizeof(buf1), &p);
439 payload_type = atoi(buf1);
440 for (i = 0; i < rt->nb_rtsp_streams; i++) {
441 rtsp_st = rt->rtsp_streams[i];
442 if (rtsp_st->sdp_payload_type == payload_type &&
443 rtsp_st->dynamic_handler &&
444 rtsp_st->dynamic_handler->parse_sdp_a_line)
445 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
446 rtsp_st->dynamic_protocol_context, buf);
448 } else if (av_strstart(p, "range:", &p)) {
451 // this is so that seeking on a streamed file can work.
452 rtsp_parse_range_npt(p, &start, &end);
453 s->start_time = start;
454 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
455 s->duration = (end == AV_NOPTS_VALUE) ?
456 AV_NOPTS_VALUE : end - start;
457 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
459 rt->transport = RTSP_TRANSPORT_RDT;
460 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
462 st = s->streams[s->nb_streams - 1];
463 st->codec->sample_rate = atoi(p);
465 if (rt->server_type == RTSP_SERVER_WMS)
466 ff_wms_parse_sdp_a_line(s, p);
467 if (s->nb_streams > 0) {
468 if (rt->server_type == RTSP_SERVER_REAL)
469 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
471 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
472 if (rtsp_st->dynamic_handler &&
473 rtsp_st->dynamic_handler->parse_sdp_a_line)
474 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
476 rtsp_st->dynamic_protocol_context, buf);
483 int ff_sdp_parse(AVFormatContext *s, const char *content)
485 RTSPState *rt = s->priv_data;
488 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
489 * contain long SDP lines containing complete ASF Headers (several
490 * kB) or arrays of MDPR (RM stream descriptor) headers plus
491 * "rulebooks" describing their properties. Therefore, the SDP line
494 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
495 * in rtpdec_xiph.c. */
497 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
499 memset(s1, 0, sizeof(SDPParseState));
502 p += strspn(p, SPACE_CHARS);
510 /* get the content */
512 while (*p != '\n' && *p != '\r' && *p != '\0') {
513 if ((q - buf) < sizeof(buf) - 1)
518 sdp_parse_line(s, s1, letter, buf);
520 while (*p != '\n' && *p != '\0')
525 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
526 if (!rt->p) return AVERROR(ENOMEM);
529 #endif /* CONFIG_RTPDEC */
531 void ff_rtsp_undo_setup(AVFormatContext *s)
533 RTSPState *rt = s->priv_data;
536 for (i = 0; i < rt->nb_rtsp_streams; i++) {
537 RTSPStream *rtsp_st = rt->rtsp_streams[i];
540 if (rtsp_st->transport_priv) {
542 AVFormatContext *rtpctx = rtsp_st->transport_priv;
543 av_write_trailer(rtpctx);
544 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
546 avio_close_dyn_buf(rtpctx->pb, &ptr);
549 avio_close(rtpctx->pb);
551 avformat_free_context(rtpctx);
552 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
553 ff_rdt_parse_close(rtsp_st->transport_priv);
554 else if (CONFIG_RTPDEC)
555 ff_rtp_parse_close(rtsp_st->transport_priv);
557 rtsp_st->transport_priv = NULL;
558 if (rtsp_st->rtp_handle)
559 ffurl_close(rtsp_st->rtp_handle);
560 rtsp_st->rtp_handle = NULL;
564 /* close and free RTSP streams */
565 void ff_rtsp_close_streams(AVFormatContext *s)
567 RTSPState *rt = s->priv_data;
571 ff_rtsp_undo_setup(s);
572 for (i = 0; i < rt->nb_rtsp_streams; i++) {
573 rtsp_st = rt->rtsp_streams[i];
575 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
576 rtsp_st->dynamic_handler->free(
577 rtsp_st->dynamic_protocol_context);
581 av_free(rt->rtsp_streams);
583 av_close_input_stream (rt->asf_ctx);
587 av_free(rt->recvbuf);
590 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
592 RTSPState *rt = s->priv_data;
595 /* open the RTP context */
596 if (rtsp_st->stream_index >= 0)
597 st = s->streams[rtsp_st->stream_index];
599 s->ctx_flags |= AVFMTCTX_NOHEADER;
601 if (s->oformat && CONFIG_RTSP_MUXER) {
602 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
604 RTSP_TCP_MAX_PACKET_SIZE);
605 /* Ownership of rtp_handle is passed to the rtp mux context */
606 rtsp_st->rtp_handle = NULL;
607 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
608 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
609 rtsp_st->dynamic_protocol_context,
610 rtsp_st->dynamic_handler);
611 else if (CONFIG_RTPDEC)
612 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
613 rtsp_st->sdp_payload_type,
614 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
615 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
617 if (!rtsp_st->transport_priv) {
618 return AVERROR(ENOMEM);
619 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
620 if (rtsp_st->dynamic_handler) {
621 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
622 rtsp_st->dynamic_protocol_context,
623 rtsp_st->dynamic_handler);
630 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
631 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
637 p += strspn(p, SPACE_CHARS);
638 v = strtol(p, (char **)&p, 10);
642 v = strtol(p, (char **)&p, 10);
651 /* XXX: only one transport specification is parsed */
652 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
654 char transport_protocol[16];
656 char lower_transport[16];
658 RTSPTransportField *th;
661 reply->nb_transports = 0;
664 p += strspn(p, SPACE_CHARS);
668 th = &reply->transports[reply->nb_transports];
670 get_word_sep(transport_protocol, sizeof(transport_protocol),
672 if (!av_strcasecmp (transport_protocol, "rtp")) {
673 get_word_sep(profile, sizeof(profile), "/;,", &p);
674 lower_transport[0] = '\0';
675 /* rtp/avp/<protocol> */
677 get_word_sep(lower_transport, sizeof(lower_transport),
680 th->transport = RTSP_TRANSPORT_RTP;
681 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
682 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
683 /* x-pn-tng/<protocol> */
684 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
686 th->transport = RTSP_TRANSPORT_RDT;
688 if (!av_strcasecmp(lower_transport, "TCP"))
689 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
691 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
695 /* get each parameter */
696 while (*p != '\0' && *p != ',') {
697 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
698 if (!strcmp(parameter, "port")) {
701 rtsp_parse_range(&th->port_min, &th->port_max, &p);
703 } else if (!strcmp(parameter, "client_port")) {
706 rtsp_parse_range(&th->client_port_min,
707 &th->client_port_max, &p);
709 } else if (!strcmp(parameter, "server_port")) {
712 rtsp_parse_range(&th->server_port_min,
713 &th->server_port_max, &p);
715 } else if (!strcmp(parameter, "interleaved")) {
718 rtsp_parse_range(&th->interleaved_min,
719 &th->interleaved_max, &p);
721 } else if (!strcmp(parameter, "multicast")) {
722 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
723 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
724 } else if (!strcmp(parameter, "ttl")) {
727 th->ttl = strtol(p, (char **)&p, 10);
729 } else if (!strcmp(parameter, "destination")) {
732 get_word_sep(buf, sizeof(buf), ";,", &p);
733 get_sockaddr(buf, &th->destination);
735 } else if (!strcmp(parameter, "source")) {
738 get_word_sep(buf, sizeof(buf), ";,", &p);
739 av_strlcpy(th->source, buf, sizeof(th->source));
743 while (*p != ';' && *p != '\0' && *p != ',')
751 reply->nb_transports++;
755 static void handle_rtp_info(RTSPState *rt, const char *url,
756 uint32_t seq, uint32_t rtptime)
759 if (!rtptime || !url[0])
761 if (rt->transport != RTSP_TRANSPORT_RTP)
763 for (i = 0; i < rt->nb_rtsp_streams; i++) {
764 RTSPStream *rtsp_st = rt->rtsp_streams[i];
765 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
768 if (!strcmp(rtsp_st->control_url, url)) {
769 rtpctx->base_timestamp = rtptime;
775 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
778 char key[20], value[1024], url[1024] = "";
779 uint32_t seq = 0, rtptime = 0;
782 p += strspn(p, SPACE_CHARS);
785 get_word_sep(key, sizeof(key), "=", &p);
789 get_word_sep(value, sizeof(value), ";, ", &p);
791 if (!strcmp(key, "url"))
792 av_strlcpy(url, value, sizeof(url));
793 else if (!strcmp(key, "seq"))
794 seq = strtoul(value, NULL, 10);
795 else if (!strcmp(key, "rtptime"))
796 rtptime = strtoul(value, NULL, 10);
798 handle_rtp_info(rt, url, seq, rtptime);
807 handle_rtp_info(rt, url, seq, rtptime);
810 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
811 RTSPState *rt, const char *method)
815 /* NOTE: we do case independent match for broken servers */
817 if (av_stristart(p, "Session:", &p)) {
819 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
820 if (av_stristart(p, ";timeout=", &p) &&
821 (t = strtol(p, NULL, 10)) > 0) {
824 } else if (av_stristart(p, "Content-Length:", &p)) {
825 reply->content_length = strtol(p, NULL, 10);
826 } else if (av_stristart(p, "Transport:", &p)) {
827 rtsp_parse_transport(reply, p);
828 } else if (av_stristart(p, "CSeq:", &p)) {
829 reply->seq = strtol(p, NULL, 10);
830 } else if (av_stristart(p, "Range:", &p)) {
831 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
832 } else if (av_stristart(p, "RealChallenge1:", &p)) {
833 p += strspn(p, SPACE_CHARS);
834 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
835 } else if (av_stristart(p, "Server:", &p)) {
836 p += strspn(p, SPACE_CHARS);
837 av_strlcpy(reply->server, p, sizeof(reply->server));
838 } else if (av_stristart(p, "Notice:", &p) ||
839 av_stristart(p, "X-Notice:", &p)) {
840 reply->notice = strtol(p, NULL, 10);
841 } else if (av_stristart(p, "Location:", &p)) {
842 p += strspn(p, SPACE_CHARS);
843 av_strlcpy(reply->location, p , sizeof(reply->location));
844 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
845 p += strspn(p, SPACE_CHARS);
846 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
847 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
848 p += strspn(p, SPACE_CHARS);
849 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
850 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
851 p += strspn(p, SPACE_CHARS);
852 if (method && !strcmp(method, "DESCRIBE"))
853 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
854 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
855 p += strspn(p, SPACE_CHARS);
856 if (method && !strcmp(method, "PLAY"))
857 rtsp_parse_rtp_info(rt, p);
858 } else if (av_stristart(p, "Public:", &p) && rt) {
859 if (strstr(p, "GET_PARAMETER") &&
860 method && !strcmp(method, "OPTIONS"))
861 rt->get_parameter_supported = 1;
862 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
863 p += strspn(p, SPACE_CHARS);
864 rt->accept_dynamic_rate = atoi(p);
868 /* skip a RTP/TCP interleaved packet */
869 void ff_rtsp_skip_packet(AVFormatContext *s)
871 RTSPState *rt = s->priv_data;
875 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
878 len = AV_RB16(buf + 1);
880 av_dlog(s, "skipping RTP packet len=%d\n", len);
885 if (len1 > sizeof(buf))
887 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
894 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
895 unsigned char **content_ptr,
896 int return_on_interleaved_data, const char *method)
898 RTSPState *rt = s->priv_data;
899 char buf[4096], buf1[1024], *q;
902 int ret, content_length, line_count = 0;
903 unsigned char *content = NULL;
905 memset(reply, 0, sizeof(*reply));
907 /* parse reply (XXX: use buffers) */
908 rt->last_reply[0] = '\0';
912 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
913 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
919 /* XXX: only parse it if first char on line ? */
920 if (return_on_interleaved_data) {
923 ff_rtsp_skip_packet(s);
924 } else if (ch != '\r') {
925 if ((q - buf) < sizeof(buf) - 1)
931 av_dlog(s, "line='%s'\n", buf);
933 /* test if last line */
937 if (line_count == 0) {
939 get_word(buf1, sizeof(buf1), &p);
940 get_word(buf1, sizeof(buf1), &p);
941 reply->status_code = atoi(buf1);
942 av_strlcpy(reply->reason, p, sizeof(reply->reason));
944 ff_rtsp_parse_line(reply, p, rt, method);
945 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
946 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
951 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
952 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
954 content_length = reply->content_length;
955 if (content_length > 0) {
956 /* leave some room for a trailing '\0' (useful for simple parsing) */
957 content = av_malloc(content_length + 1);
958 ffurl_read_complete(rt->rtsp_hd, content, content_length);
959 content[content_length] = '\0';
962 *content_ptr = content;
966 if (rt->seq != reply->seq) {
967 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
968 rt->seq, reply->seq);
972 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
973 reply->notice == 2104 /* Start-of-Stream Reached */ ||
974 reply->notice == 2306 /* Continuous Feed Terminated */) {
975 rt->state = RTSP_STATE_IDLE;
976 } else if (reply->notice >= 4400 && reply->notice < 5500) {
977 return AVERROR(EIO); /* data or server error */
978 } else if (reply->notice == 2401 /* Ticket Expired */ ||
979 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
980 return AVERROR(EPERM);
986 * Send a command to the RTSP server without waiting for the reply.
988 * @param s RTSP (de)muxer context
989 * @param method the method for the request
990 * @param url the target url for the request
991 * @param headers extra header lines to include in the request
992 * @param send_content if non-null, the data to send as request body content
993 * @param send_content_length the length of the send_content data, or 0 if
994 * send_content is null
996 * @return zero if success, nonzero otherwise
998 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
999 const char *method, const char *url,
1000 const char *headers,
1001 const unsigned char *send_content,
1002 int send_content_length)
1004 RTSPState *rt = s->priv_data;
1005 char buf[4096], *out_buf;
1006 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1008 /* Add in RTSP headers */
1011 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1013 av_strlcat(buf, headers, sizeof(buf));
1014 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1015 if (rt->session_id[0] != '\0' && (!headers ||
1016 !strstr(headers, "\nIf-Match:"))) {
1017 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1020 char *str = ff_http_auth_create_response(&rt->auth_state,
1021 rt->auth, url, method);
1023 av_strlcat(buf, str, sizeof(buf));
1026 if (send_content_length > 0 && send_content)
1027 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1028 av_strlcat(buf, "\r\n", sizeof(buf));
1030 /* base64 encode rtsp if tunneling */
1031 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1032 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1033 out_buf = base64buf;
1036 av_dlog(s, "Sending:\n%s--\n", buf);
1038 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1039 if (send_content_length > 0 && send_content) {
1040 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1041 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1042 "with content data not supported\n");
1043 return AVERROR_PATCHWELCOME;
1045 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1047 rt->last_cmd_time = av_gettime();
1052 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1053 const char *url, const char *headers)
1055 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1058 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1059 const char *headers, RTSPMessageHeader *reply,
1060 unsigned char **content_ptr)
1062 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1063 content_ptr, NULL, 0);
1066 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1067 const char *method, const char *url,
1069 RTSPMessageHeader *reply,
1070 unsigned char **content_ptr,
1071 const unsigned char *send_content,
1072 int send_content_length)
1074 RTSPState *rt = s->priv_data;
1075 HTTPAuthType cur_auth_type;
1079 cur_auth_type = rt->auth_state.auth_type;
1080 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1082 send_content_length)))
1085 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1088 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1089 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1092 if (reply->status_code > 400){
1093 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1097 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1103 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1104 int lower_transport, const char *real_challenge)
1106 RTSPState *rt = s->priv_data;
1107 int rtx, j, i, err, interleave = 0;
1108 RTSPStream *rtsp_st;
1109 RTSPMessageHeader reply1, *reply = &reply1;
1111 const char *trans_pref;
1113 if (rt->transport == RTSP_TRANSPORT_RDT)
1114 trans_pref = "x-pn-tng";
1116 trans_pref = "RTP/AVP";
1118 /* default timeout: 1 minute */
1121 /* for each stream, make the setup request */
1122 /* XXX: we assume the same server is used for the control of each
1125 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1126 char transport[2048];
1129 * WMS serves all UDP data over a single connection, the RTX, which
1130 * isn't necessarily the first in the SDP but has to be the first
1131 * to be set up, else the second/third SETUP will fail with a 461.
1133 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1134 rt->server_type == RTSP_SERVER_WMS) {
1137 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1138 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1140 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1144 if (rtx == rt->nb_rtsp_streams)
1145 return -1; /* no RTX found */
1146 rtsp_st = rt->rtsp_streams[rtx];
1148 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1150 rtsp_st = rt->rtsp_streams[i];
1153 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1156 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1157 port = reply->transports[0].client_port_min;
1161 /* first try in specified port range */
1162 if (RTSP_RTP_PORT_MIN != 0) {
1163 while (j <= RTSP_RTP_PORT_MAX) {
1164 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1165 "?localport=%d", j);
1166 /* we will use two ports per rtp stream (rtp and rtcp) */
1168 if (ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1169 &s->interrupt_callback, NULL) == 0)
1174 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1179 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1181 snprintf(transport, sizeof(transport) - 1,
1182 "%s/UDP;", trans_pref);
1183 if (rt->server_type != RTSP_SERVER_REAL)
1184 av_strlcat(transport, "unicast;", sizeof(transport));
1185 av_strlcatf(transport, sizeof(transport),
1186 "client_port=%d", port);
1187 if (rt->transport == RTSP_TRANSPORT_RTP &&
1188 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1189 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1193 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1194 /* For WMS streams, the application streams are only used for
1195 * UDP. When trying to set it up for TCP streams, the server
1196 * will return an error. Therefore, we skip those streams. */
1197 if (rt->server_type == RTSP_SERVER_WMS &&
1198 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1201 snprintf(transport, sizeof(transport) - 1,
1202 "%s/TCP;", trans_pref);
1203 if (rt->transport != RTSP_TRANSPORT_RDT)
1204 av_strlcat(transport, "unicast;", sizeof(transport));
1205 av_strlcatf(transport, sizeof(transport),
1206 "interleaved=%d-%d",
1207 interleave, interleave + 1);
1211 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1212 snprintf(transport, sizeof(transport) - 1,
1213 "%s/UDP;multicast", trans_pref);
1216 av_strlcat(transport, ";mode=receive", sizeof(transport));
1217 } else if (rt->server_type == RTSP_SERVER_REAL ||
1218 rt->server_type == RTSP_SERVER_WMS)
1219 av_strlcat(transport, ";mode=play", sizeof(transport));
1220 snprintf(cmd, sizeof(cmd),
1221 "Transport: %s\r\n",
1223 if (rt->accept_dynamic_rate)
1224 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1225 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1226 char real_res[41], real_csum[9];
1227 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1229 av_strlcatf(cmd, sizeof(cmd),
1231 "RealChallenge2: %s, sd=%s\r\n",
1232 rt->session_id, real_res, real_csum);
1234 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1235 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1238 } else if (reply->status_code != RTSP_STATUS_OK ||
1239 reply->nb_transports != 1) {
1240 err = AVERROR_INVALIDDATA;
1244 /* XXX: same protocol for all streams is required */
1246 if (reply->transports[0].lower_transport != rt->lower_transport ||
1247 reply->transports[0].transport != rt->transport) {
1248 err = AVERROR_INVALIDDATA;
1252 rt->lower_transport = reply->transports[0].lower_transport;
1253 rt->transport = reply->transports[0].transport;
1256 /* Fail if the server responded with another lower transport mode
1257 * than what we requested. */
1258 if (reply->transports[0].lower_transport != lower_transport) {
1259 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1260 err = AVERROR_INVALIDDATA;
1264 switch(reply->transports[0].lower_transport) {
1265 case RTSP_LOWER_TRANSPORT_TCP:
1266 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1267 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1270 case RTSP_LOWER_TRANSPORT_UDP: {
1271 char url[1024], options[30] = "";
1273 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1274 av_strlcpy(options, "?connect=1", sizeof(options));
1275 /* Use source address if specified */
1276 if (reply->transports[0].source[0]) {
1277 ff_url_join(url, sizeof(url), "rtp", NULL,
1278 reply->transports[0].source,
1279 reply->transports[0].server_port_min, "%s", options);
1281 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1282 reply->transports[0].server_port_min, "%s", options);
1284 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1285 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1286 err = AVERROR_INVALIDDATA;
1289 /* Try to initialize the connection state in a
1290 * potential NAT router by sending dummy packets.
1291 * RTP/RTCP dummy packets are used for RDT, too.
1293 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1295 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1298 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1299 char url[1024], namebuf[50];
1300 struct sockaddr_storage addr;
1303 if (reply->transports[0].destination.ss_family) {
1304 addr = reply->transports[0].destination;
1305 port = reply->transports[0].port_min;
1306 ttl = reply->transports[0].ttl;
1308 addr = rtsp_st->sdp_ip;
1309 port = rtsp_st->sdp_port;
1310 ttl = rtsp_st->sdp_ttl;
1312 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1313 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1314 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1315 port, "?ttl=%d", ttl);
1316 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1317 &s->interrupt_callback, NULL) < 0) {
1318 err = AVERROR_INVALIDDATA;
1325 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1329 if (reply->timeout > 0)
1330 rt->timeout = reply->timeout;
1332 if (rt->server_type == RTSP_SERVER_REAL)
1333 rt->need_subscription = 1;
1338 ff_rtsp_undo_setup(s);
1342 void ff_rtsp_close_connections(AVFormatContext *s)
1344 RTSPState *rt = s->priv_data;
1345 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1346 ffurl_close(rt->rtsp_hd);
1347 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1350 int ff_rtsp_connect(AVFormatContext *s)
1352 RTSPState *rt = s->priv_data;
1353 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1354 char *option_list, *option, *filename;
1355 int port, err, tcp_fd;
1356 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1357 int lower_transport_mask = 0;
1358 char real_challenge[64] = "";
1359 struct sockaddr_storage peer;
1360 socklen_t peer_len = sizeof(peer);
1362 if (!ff_network_init())
1363 return AVERROR(EIO);
1365 rt->control_transport = RTSP_MODE_PLAIN;
1366 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1367 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1368 rt->control_transport = RTSP_MODE_TUNNEL;
1370 /* Only pass through valid flags from here */
1371 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1374 lower_transport_mask = rt->lower_transport_mask;
1375 /* extract hostname and port */
1376 av_url_split(NULL, 0, auth, sizeof(auth),
1377 host, sizeof(host), &port, path, sizeof(path), s->filename);
1379 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1382 port = RTSP_DEFAULT_PORT;
1384 #if FF_API_RTSP_URL_OPTIONS
1385 /* search for options */
1386 option_list = strrchr(path, '?');
1388 /* Strip out the RTSP specific options, write out the rest of
1389 * the options back into the same string. */
1390 filename = option_list;
1391 while (option_list) {
1393 /* move the option pointer */
1394 option = ++option_list;
1395 option_list = strchr(option_list, '&');
1399 /* handle the options */
1400 if (!strcmp(option, "udp")) {
1401 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1402 } else if (!strcmp(option, "multicast")) {
1403 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1404 } else if (!strcmp(option, "tcp")) {
1405 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1406 } else if(!strcmp(option, "http")) {
1407 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1408 rt->control_transport = RTSP_MODE_TUNNEL;
1409 } else if (!strcmp(option, "filter_src")) {
1410 rt->rtsp_flags |= RTSP_FLAG_FILTER_SRC;
1412 /* Write options back into the buffer, using memmove instead
1413 * of strcpy since the strings may overlap. */
1414 int len = strlen(option);
1415 memmove(++filename, option, len);
1417 if (option_list) *filename = '&';
1421 av_log(s, AV_LOG_WARNING, "Options passed via URL are "
1422 "deprecated, use -rtsp_transport "
1423 "and -rtsp_flags instead.\n");
1429 if (!lower_transport_mask)
1430 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1433 /* Only UDP or TCP - UDP multicast isn't supported. */
1434 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1435 (1 << RTSP_LOWER_TRANSPORT_TCP);
1436 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1437 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1438 "only UDP and TCP are supported for output.\n");
1439 err = AVERROR(EINVAL);
1444 /* Construct the URI used in request; this is similar to s->filename,
1445 * but with authentication credentials removed and RTSP specific options
1447 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1448 host, port, "%s", path);
1450 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1451 /* set up initial handshake for tunneling */
1452 char httpname[1024];
1453 char sessioncookie[17];
1456 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1457 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1458 av_get_random_seed(), av_get_random_seed());
1461 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1462 &s->interrupt_callback) < 0) {
1467 /* generate GET headers */
1468 snprintf(headers, sizeof(headers),
1469 "x-sessioncookie: %s\r\n"
1470 "Accept: application/x-rtsp-tunnelled\r\n"
1471 "Pragma: no-cache\r\n"
1472 "Cache-Control: no-cache\r\n",
1474 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1476 /* complete the connection */
1477 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1483 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1484 &s->interrupt_callback) < 0 ) {
1489 /* generate POST headers */
1490 snprintf(headers, sizeof(headers),
1491 "x-sessioncookie: %s\r\n"
1492 "Content-Type: application/x-rtsp-tunnelled\r\n"
1493 "Pragma: no-cache\r\n"
1494 "Cache-Control: no-cache\r\n"
1495 "Content-Length: 32767\r\n"
1496 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1498 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1499 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1501 /* Initialize the authentication state for the POST session. The HTTP
1502 * protocol implementation doesn't properly handle multi-pass
1503 * authentication for POST requests, since it would require one of
1505 * - implementing Expect: 100-continue, which many HTTP servers
1506 * don't support anyway, even less the RTSP servers that do HTTP
1508 * - sending the whole POST data until getting a 401 reply specifying
1509 * what authentication method to use, then resending all that data
1510 * - waiting for potential 401 replies directly after sending the
1511 * POST header (waiting for some unspecified time)
1512 * Therefore, we copy the full auth state, which works for both basic
1513 * and digest. (For digest, we would have to synchronize the nonce
1514 * count variable between the two sessions, if we'd do more requests
1515 * with the original session, though.)
1517 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1519 /* complete the connection */
1520 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1525 /* open the tcp connection */
1526 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1527 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1528 &s->interrupt_callback, NULL) < 0) {
1532 rt->rtsp_hd_out = rt->rtsp_hd;
1536 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1537 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1538 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1539 NULL, 0, NI_NUMERICHOST);
1542 /* request options supported by the server; this also detects server
1544 for (rt->server_type = RTSP_SERVER_RTP;;) {
1546 if (rt->server_type == RTSP_SERVER_REAL)
1549 * The following entries are required for proper
1550 * streaming from a Realmedia server. They are
1551 * interdependent in some way although we currently
1552 * don't quite understand how. Values were copied
1553 * from mplayer SVN r23589.
1554 * ClientChallenge is a 16-byte ID in hex
1555 * CompanyID is a 16-byte ID in base64
1557 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1558 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1559 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1560 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1562 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1563 if (reply->status_code != RTSP_STATUS_OK) {
1564 err = AVERROR_INVALIDDATA;
1568 /* detect server type if not standard-compliant RTP */
1569 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1570 rt->server_type = RTSP_SERVER_REAL;
1572 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1573 rt->server_type = RTSP_SERVER_WMS;
1574 } else if (rt->server_type == RTSP_SERVER_REAL)
1575 strcpy(real_challenge, reply->real_challenge);
1579 if (s->iformat && CONFIG_RTSP_DEMUXER)
1580 err = ff_rtsp_setup_input_streams(s, reply);
1581 else if (CONFIG_RTSP_MUXER)
1582 err = ff_rtsp_setup_output_streams(s, host);
1587 int lower_transport = ff_log2_tab[lower_transport_mask &
1588 ~(lower_transport_mask - 1)];
1590 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1591 rt->server_type == RTSP_SERVER_REAL ?
1592 real_challenge : NULL);
1595 lower_transport_mask &= ~(1 << lower_transport);
1596 if (lower_transport_mask == 0 && err == 1) {
1597 err = AVERROR(EPROTONOSUPPORT);
1602 rt->lower_transport_mask = lower_transport_mask;
1603 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1604 rt->state = RTSP_STATE_IDLE;
1605 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1608 ff_rtsp_close_streams(s);
1609 ff_rtsp_close_connections(s);
1610 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1611 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1612 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1620 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1623 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1624 uint8_t *buf, int buf_size, int64_t wait_end)
1626 RTSPState *rt = s->priv_data;
1627 RTSPStream *rtsp_st;
1628 int n, i, ret, tcp_fd, timeout_cnt = 0;
1630 struct pollfd *p = rt->p;
1633 if (ff_check_interrupt(&s->interrupt_callback))
1634 return AVERROR_EXIT;
1635 if (wait_end && wait_end - av_gettime() < 0)
1636 return AVERROR(EAGAIN);
1639 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1640 p[max_p].fd = tcp_fd;
1641 p[max_p++].events = POLLIN;
1645 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1646 rtsp_st = rt->rtsp_streams[i];
1647 if (rtsp_st->rtp_handle) {
1648 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1649 p[max_p++].events = POLLIN;
1650 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1651 p[max_p++].events = POLLIN;
1654 n = poll(p, max_p, POLL_TIMEOUT_MS);
1656 int j = 1 - (tcp_fd == -1);
1658 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1659 rtsp_st = rt->rtsp_streams[i];
1660 if (rtsp_st->rtp_handle) {
1661 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1662 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1664 *prtsp_st = rtsp_st;
1671 #if CONFIG_RTSP_DEMUXER
1672 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1673 RTSPMessageHeader reply;
1675 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1678 /* XXX: parse message */
1679 if (rt->state != RTSP_STATE_STREAMING)
1683 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1684 return AVERROR(ETIMEDOUT);
1685 } else if (n < 0 && errno != EINTR)
1686 return AVERROR(errno);
1690 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1692 RTSPState *rt = s->priv_data;
1694 RTSPStream *rtsp_st, *first_queue_st = NULL;
1695 int64_t wait_end = 0;
1697 if (rt->nb_byes == rt->nb_rtsp_streams)
1700 /* get next frames from the same RTP packet */
1701 if (rt->cur_transport_priv) {
1702 if (rt->transport == RTSP_TRANSPORT_RDT) {
1703 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1705 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1707 rt->cur_transport_priv = NULL;
1709 } else if (ret == 1) {
1712 rt->cur_transport_priv = NULL;
1715 if (rt->transport == RTSP_TRANSPORT_RTP) {
1717 int64_t first_queue_time = 0;
1718 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1719 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1723 queue_time = ff_rtp_queued_packet_time(rtpctx);
1724 if (queue_time && (queue_time - first_queue_time < 0 ||
1725 !first_queue_time)) {
1726 first_queue_time = queue_time;
1727 first_queue_st = rt->rtsp_streams[i];
1730 if (first_queue_time)
1731 wait_end = first_queue_time + s->max_delay;
1734 /* read next RTP packet */
1737 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1739 return AVERROR(ENOMEM);
1742 switch(rt->lower_transport) {
1744 #if CONFIG_RTSP_DEMUXER
1745 case RTSP_LOWER_TRANSPORT_TCP:
1746 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1749 case RTSP_LOWER_TRANSPORT_UDP:
1750 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1751 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1752 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1753 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1756 if (len == AVERROR(EAGAIN) && first_queue_st &&
1757 rt->transport == RTSP_TRANSPORT_RTP) {
1758 rtsp_st = first_queue_st;
1759 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1766 if (rt->transport == RTSP_TRANSPORT_RDT) {
1767 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1769 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1771 /* Either bad packet, or a RTCP packet. Check if the
1772 * first_rtcp_ntp_time field was initialized. */
1773 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1774 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1775 /* first_rtcp_ntp_time has been initialized for this stream,
1776 * copy the same value to all other uninitialized streams,
1777 * in order to map their timestamp origin to the same ntp time
1780 AVStream *st = NULL;
1781 if (rtsp_st->stream_index >= 0)
1782 st = s->streams[rtsp_st->stream_index];
1783 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1784 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1785 AVStream *st2 = NULL;
1786 if (rt->rtsp_streams[i]->stream_index >= 0)
1787 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1788 if (rtpctx2 && st && st2 &&
1789 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1790 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1791 rtpctx2->rtcp_ts_offset = av_rescale_q(
1792 rtpctx->rtcp_ts_offset, st->time_base,
1797 if (ret == -RTCP_BYE) {
1800 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1801 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1803 if (rt->nb_byes == rt->nb_rtsp_streams)
1812 /* more packets may follow, so we save the RTP context */
1813 rt->cur_transport_priv = rtsp_st->transport_priv;
1817 #endif /* CONFIG_RTPDEC */
1819 #if CONFIG_SDP_DEMUXER
1820 static int sdp_probe(AVProbeData *p1)
1822 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1824 /* we look for a line beginning "c=IN IP" */
1825 while (p < p_end && *p != '\0') {
1826 if (p + sizeof("c=IN IP") - 1 < p_end &&
1827 av_strstart(p, "c=IN IP", NULL))
1828 return AVPROBE_SCORE_MAX / 2;
1830 while (p < p_end - 1 && *p != '\n') p++;
1839 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1841 RTSPState *rt = s->priv_data;
1842 RTSPStream *rtsp_st;
1847 if (!ff_network_init())
1848 return AVERROR(EIO);
1850 /* read the whole sdp file */
1851 /* XXX: better loading */
1852 content = av_malloc(SDP_MAX_SIZE);
1853 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1856 return AVERROR_INVALIDDATA;
1858 content[size] ='\0';
1860 err = ff_sdp_parse(s, content);
1864 /* open each RTP stream */
1865 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1867 rtsp_st = rt->rtsp_streams[i];
1869 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1870 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1871 ff_url_join(url, sizeof(url), "rtp", NULL,
1872 namebuf, rtsp_st->sdp_port,
1873 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1875 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1876 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1877 &s->interrupt_callback, NULL) < 0) {
1878 err = AVERROR_INVALIDDATA;
1881 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1886 ff_rtsp_close_streams(s);
1891 static int sdp_read_close(AVFormatContext *s)
1893 ff_rtsp_close_streams(s);
1898 static const AVClass sdp_demuxer_class = {
1899 .class_name = "SDP demuxer",
1900 .item_name = av_default_item_name,
1901 .option = sdp_options,
1902 .version = LIBAVUTIL_VERSION_INT,
1905 AVInputFormat ff_sdp_demuxer = {
1907 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1908 .priv_data_size = sizeof(RTSPState),
1909 .read_probe = sdp_probe,
1910 .read_header = sdp_read_header,
1911 .read_packet = ff_rtsp_fetch_packet,
1912 .read_close = sdp_read_close,
1913 .priv_class = &sdp_demuxer_class
1915 #endif /* CONFIG_SDP_DEMUXER */
1917 #if CONFIG_RTP_DEMUXER
1918 static int rtp_probe(AVProbeData *p)
1920 if (av_strstart(p->filename, "rtp:", NULL))
1921 return AVPROBE_SCORE_MAX;
1925 static int rtp_read_header(AVFormatContext *s,
1926 AVFormatParameters *ap)
1928 uint8_t recvbuf[1500];
1929 char host[500], sdp[500];
1931 URLContext* in = NULL;
1933 AVCodecContext codec;
1934 struct sockaddr_storage addr;
1936 socklen_t addrlen = sizeof(addr);
1937 RTSPState *rt = s->priv_data;
1939 if (!ff_network_init())
1940 return AVERROR(EIO);
1942 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1943 &s->interrupt_callback, NULL);
1948 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1949 if (ret == AVERROR(EAGAIN))
1954 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1958 if ((recvbuf[0] & 0xc0) != 0x80) {
1959 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1964 payload_type = recvbuf[1] & 0x7f;
1967 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1971 memset(&codec, 0, sizeof(codec));
1972 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1973 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1974 "without an SDP file describing it\n",
1978 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1979 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1980 "properly you need an SDP file "
1984 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1985 NULL, 0, s->filename);
1987 snprintf(sdp, sizeof(sdp),
1988 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1989 addr.ss_family == AF_INET ? 4 : 6, host,
1990 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1991 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1992 port, payload_type);
1993 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1995 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1998 /* sdp_read_header initializes this again */
2001 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2003 ret = sdp_read_header(s, ap);
2014 static const AVClass rtp_demuxer_class = {
2015 .class_name = "RTP demuxer",
2016 .item_name = av_default_item_name,
2017 .option = rtp_options,
2018 .version = LIBAVUTIL_VERSION_INT,
2021 AVInputFormat ff_rtp_demuxer = {
2023 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2024 .priv_data_size = sizeof(RTSPState),
2025 .read_probe = rtp_probe,
2026 .read_header = rtp_read_header,
2027 .read_packet = ff_rtsp_fetch_packet,
2028 .read_close = sdp_read_close,
2029 .flags = AVFMT_NOFILE,
2030 .priv_class = &rtp_demuxer_class
2032 #endif /* CONFIG_RTP_DEMUXER */